mirror of https://github.com/mstorsjo/fdk-aac.git
Add a tool for testing decoding+encoding in a number of configurations
This commit is contained in:
Martin Storsjo
parent
2326faaf8f
commit
bf5a6867c7
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@ -39,11 +39,14 @@ libfdk_aac_la_LDFLAGS = -version-info @FDK_AAC_VERSION@ -no-undefined \
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-export-symbols $(top_srcdir)/fdk-aac.sym
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if EXAMPLE
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bin_PROGRAMS = aac-enc$(EXEEXT)
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bin_PROGRAMS = aac-enc$(EXEEXT) test-encode-decode$(EXEEXT)
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aac_enc_LDADD = libfdk-aac.la
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aac_enc_SOURCES = aac-enc.c wavreader.c
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test_encode_decode_LDADD = libfdk-aac.la
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test_encode_decode_SOURCES = test-encode-decode.c wavreader.c
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noinst_HEADERS = wavreader.h
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endif
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@ -0,0 +1,365 @@
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/* ------------------------------------------------------------------
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* Copyright (C) 2011 Martin Storsjo
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
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* express or implied.
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* See the License for the specific language governing permissions
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* and limitations under the License.
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* -------------------------------------------------------------------
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*/
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#include <stdio.h>
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#include <stdint.h>
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#include <stdlib.h>
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#include <string.h>
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#include "aacenc_lib.h"
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#include "aacdecoder_lib.h"
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#include "wavreader.h"
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static int encoder_input_samples, encoder_input_size;
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static int decoder_output_skip;
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static int16_t *encoder_input;
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static int max_diff;
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static uint64_t diff_sum, diff_samples;
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static void init_encoder_input(void) {
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encoder_input_samples = 0;
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max_diff = 0;
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diff_sum = diff_samples = 0;
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}
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static void free_encoder_input(void) {
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free(encoder_input);
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encoder_input = NULL;
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encoder_input_size = 0;
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}
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static void append_encoder_input(const int16_t *input, int samples) {
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if (encoder_input_samples + samples > encoder_input_size) {
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int size = 2*(encoder_input_samples + samples);
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int16_t *ptr = realloc(encoder_input, size * sizeof(*encoder_input));
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if (!ptr)
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abort();
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encoder_input = ptr;
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encoder_input_size = size;
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}
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memcpy(encoder_input + encoder_input_samples, input, samples * sizeof(*input));
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encoder_input_samples += samples;
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}
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static void compare_decoder_output(const int16_t *output, int samples) {
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int i;
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// TODO: Stereo upconvert?
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if (decoder_output_skip > 0) {
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int n = samples;
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if (n > decoder_output_skip)
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n = decoder_output_skip;
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output += n;
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samples -= n;
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decoder_output_skip -= n;
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if (samples <= 0)
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return;
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}
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if (samples > encoder_input_samples)
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samples = encoder_input_samples;
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for (i = 0; i < samples; i++) {
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int diff = abs(encoder_input[i] - output[i]);
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if (diff > max_diff)
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max_diff = diff;
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diff_sum += diff;
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diff_samples++;
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}
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memmove(encoder_input, encoder_input + samples, (encoder_input_samples - samples) * sizeof(*encoder_input));
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encoder_input_samples -= samples;
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}
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static int decode(HANDLE_AACDECODER decoder, const uint8_t *ptr, int size, uint8_t *decoder_buffer, int decoder_buffer_size, int channels) {
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AAC_DECODER_ERROR err;
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CStreamInfo *info;
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UINT valid, buffer_size;
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do {
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valid = buffer_size = size;
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err = aacDecoder_Fill(decoder, (UCHAR**) &ptr, &buffer_size, &valid);
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ptr += buffer_size - valid;
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size -= buffer_size - valid;
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if (err == AAC_DEC_NOT_ENOUGH_BITS)
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continue;
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if (err != AAC_DEC_OK)
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break;
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err = aacDecoder_DecodeFrame(decoder, (INT_PCM *) decoder_buffer, decoder_buffer_size / sizeof(INT_PCM), 0);
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if (!ptr && err != AAC_DEC_OK)
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break;
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if (err == AAC_DEC_NOT_ENOUGH_BITS)
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continue;
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if (err != AAC_DEC_OK) {
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fprintf(stderr, "Decoding failed\n");
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return 1;
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}
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info = aacDecoder_GetStreamInfo(decoder);
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if (info->numChannels != channels) {
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fprintf(stderr, "Mismatched number of channels, input %d, output %d\n", channels, info->numChannels);
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return 1;
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}
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compare_decoder_output((int16_t*) decoder_buffer, info->numChannels * info->frameSize);
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} while (size > 0);
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return 0;
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}
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static int test_encode_decode(const char *infile, int aot, int afterburner, int eld_sbr, int vbr, int bitrate, int adts) {
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void *wav;
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int format, sample_rate, channels, bits_per_sample;
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int input_size;
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uint8_t* input_buf;
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int16_t* convert_buf;
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HANDLE_AACENCODER encoder;
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CHANNEL_MODE mode;
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AACENC_InfoStruct info = { 0 };
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HANDLE_AACDECODER decoder;
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int ret = 0;
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int decoder_buffer_size = 2048 * 2 * 8;
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uint8_t* decoder_buffer = malloc(decoder_buffer_size);
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int avg_diff;
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printf("Testing encoding with aot %d afterburner %d eld_sbr %d vbr %d bitrate %d adts %d\n", aot, afterburner, eld_sbr, vbr, bitrate, adts);
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init_encoder_input();
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wav = wav_read_open(infile);
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if (!wav) {
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fprintf(stderr, "Unable to open wav file %s\n", infile);
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return 1;
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}
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if (!wav_get_header(wav, &format, &channels, &sample_rate, &bits_per_sample, NULL)) {
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fprintf(stderr, "Bad wav file %s\n", infile);
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return 1;
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}
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if (format != 1) {
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fprintf(stderr, "Unsupported WAV format %d\n", format);
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return 1;
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}
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if (bits_per_sample != 16) {
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fprintf(stderr, "Unsupported WAV sample depth %d\n", bits_per_sample);
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return 1;
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}
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switch (channels) {
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case 1: mode = MODE_1; break;
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case 2: mode = MODE_2; break;
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case 3: mode = MODE_1_2; break;
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case 4: mode = MODE_1_2_1; break;
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case 5: mode = MODE_1_2_2; break;
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case 6: mode = MODE_1_2_2_1; break;
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default:
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fprintf(stderr, "Unsupported WAV channels %d\n", channels);
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return 1;
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}
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if (aacEncOpen(&encoder, 0, channels) != AACENC_OK) {
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fprintf(stderr, "Unable to open encoder\n");
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return 1;
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}
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if (aacEncoder_SetParam(encoder, AACENC_AOT, aot) != AACENC_OK) {
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fprintf(stderr, "Unable to set the AOT\n");
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return 1;
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}
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if (aot == 39 && eld_sbr) {
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if (aacEncoder_SetParam(encoder, AACENC_SBR_MODE, 1) != AACENC_OK) {
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fprintf(stderr, "Unable to set SBR mode for ELD\n");
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return 1;
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}
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}
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if (aacEncoder_SetParam(encoder, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
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fprintf(stderr, "Unable to set the AOT\n");
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return 1;
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}
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if (aacEncoder_SetParam(encoder, AACENC_CHANNELMODE, mode) != AACENC_OK) {
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fprintf(stderr, "Unable to set the channel mode\n");
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return 1;
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}
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if (aacEncoder_SetParam(encoder, AACENC_CHANNELORDER, 1) != AACENC_OK) {
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fprintf(stderr, "Unable to set the wav channel order\n");
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return 1;
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}
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if (vbr) {
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if (aacEncoder_SetParam(encoder, AACENC_BITRATEMODE, vbr) != AACENC_OK) {
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fprintf(stderr, "Unable to set the VBR bitrate mode\n");
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return 1;
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}
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} else {
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if (aacEncoder_SetParam(encoder, AACENC_BITRATE, bitrate) != AACENC_OK) {
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fprintf(stderr, "Unable to set the bitrate\n");
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return 1;
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}
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}
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if (aacEncoder_SetParam(encoder, AACENC_TRANSMUX, adts ? 2 : 0) != AACENC_OK) {
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fprintf(stderr, "Unable to set the ADTS transmux\n");
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return 1;
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}
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if (aacEncoder_SetParam(encoder, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
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fprintf(stderr, "Unable to set the afterburner mode\n");
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return 1;
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}
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if (aacEncEncode(encoder, NULL, NULL, NULL, NULL) != AACENC_OK) {
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fprintf(stderr, "Unable to initialize the encoder\n");
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return 1;
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}
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if (aacEncInfo(encoder, &info) != AACENC_OK) {
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fprintf(stderr, "Unable to get the encoder info\n");
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return 1;
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}
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input_size = channels*2*info.frameLength;
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input_buf = (uint8_t*) malloc(input_size);
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convert_buf = (int16_t*) malloc(input_size);
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decoder_output_skip = channels * info.nDelay;
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decoder = aacDecoder_Open(adts ? TT_MP4_ADTS : TT_MP4_RAW, 1);
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if (!adts) {
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UCHAR *bufArray[] = { info.confBuf };
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if (aacDecoder_ConfigRaw(decoder, (UCHAR**) bufArray, &info.confSize) != AAC_DEC_OK) {
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fprintf(stderr, "Unable to set ASC\n");
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ret = 1;
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goto end;
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}
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}
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aacDecoder_SetParam(decoder, AAC_CONCEAL_METHOD, 1);
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aacDecoder_SetParam(decoder, AAC_PCM_LIMITER_ENABLE, 0);
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while (1) {
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AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
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AACENC_InArgs in_args = { 0 };
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AACENC_OutArgs out_args = { 0 };
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int in_identifier = IN_AUDIO_DATA;
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int in_size, in_elem_size;
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int out_identifier = OUT_BITSTREAM_DATA;
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int out_size, out_elem_size;
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int read, i;
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void *in_ptr, *out_ptr;
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uint8_t outbuf[20480];
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AACENC_ERROR err;
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read = wav_read_data(wav, input_buf, input_size);
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for (i = 0; i < read/2; i++) {
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const uint8_t* in = &input_buf[2*i];
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convert_buf[i] = in[0] | (in[1] << 8);
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}
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in_ptr = convert_buf;
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in_size = read;
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in_elem_size = 2;
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in_buf.numBufs = 1;
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in_buf.bufs = &in_ptr;
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in_buf.bufferIdentifiers = &in_identifier;
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in_buf.bufSizes = &in_size;
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in_buf.bufElSizes = &in_elem_size;
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if (read <= 0) {
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in_args.numInSamples = -1;
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} else {
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in_args.numInSamples = read/2;
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append_encoder_input(convert_buf, in_args.numInSamples);
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}
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out_ptr = outbuf;
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out_size = sizeof(outbuf);
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out_elem_size = 1;
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out_buf.numBufs = 1;
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out_buf.bufs = &out_ptr;
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out_buf.bufferIdentifiers = &out_identifier;
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out_buf.bufSizes = &out_size;
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out_buf.bufElSizes = &out_elem_size;
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if ((err = aacEncEncode(encoder, &in_buf, &out_buf, &in_args, &out_args)) != AACENC_OK) {
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if (err == AACENC_ENCODE_EOF)
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break;
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fprintf(stderr, "Encoding failed\n");
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ret = 1;
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goto end;
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}
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if (out_args.numOutBytes == 0)
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continue;
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if (decode(decoder, outbuf, out_args.numOutBytes, decoder_buffer, decoder_buffer_size, channels)) {
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ret = 1;
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goto end;
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}
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}
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if (encoder_input_samples > 0) {
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fprintf(stderr, "%d unmatched samples left at the end\n", encoder_input_samples);
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ret = 1;
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goto end;
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}
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avg_diff = 0;
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if (diff_samples > 0)
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avg_diff = diff_sum / diff_samples;
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if (/*max_diff > 7000 ||*/ avg_diff > ((aot == 23) ? 2500 : (aot == 29) ? 1000 : 300)) {
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fprintf(stderr, "max_diff %d, avg_diff %d\n", max_diff, avg_diff);
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ret = 1;
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goto end;
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}
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end:
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free(input_buf);
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free(convert_buf);
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wav_read_close(wav);
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aacEncClose(&encoder);
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free(decoder_buffer);
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aacDecoder_Close(decoder);
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return ret;
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}
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int main(int argc, char *argv[]) {
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const char* infile;
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void *wav;
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int sample_rate, channels;
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int failures = 0;
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int i;
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if (argc < 2) {
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printf("%s input.wav\n", argv[0]);
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return 1;
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}
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infile = argv[1];
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wav = wav_read_open(infile);
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if (!wav) {
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fprintf(stderr, "Unable to open wav file %s\n", infile);
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return 1;
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}
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if (!wav_get_header(wav, NULL, &channels, &sample_rate, NULL, NULL)) {
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fprintf(stderr, "Bad wav file %s\n", infile);
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return 1;
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}
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wav_read_close(wav);
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failures += test_encode_decode(infile, 2, 0, 0, 0, 64000, 0); // AAC-LC, without afterburner
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for (i = 0; i < 2; i++)
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failures += test_encode_decode(infile, 2, 1, 0, 0, 64000, i); // AAC-LC
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for (i = 1; i <= 5; i++)
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failures += test_encode_decode(infile, 2, 1, 0, i, 0, 0); // AAC-LC VBR
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if (channels == 2) {
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// HE-AACv2 only works for stereo; HE-AACv1 gets upconverted to stereo (which we don't match properly)
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for (i = 0; i < 2; i++)
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failures += test_encode_decode(infile, 5, 1, 0, 0, 64000, i); // HE-AAC
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for (i = 1; i <= 5; i++)
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failures += test_encode_decode(infile, 5, 1, 0, i, 0, 0); // HE-AAC VBR
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for (i = 0; i < 2; i++)
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failures += test_encode_decode(infile, 29, 1, 0, 0, 64000, i); // HE-AACv2
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for (i = 1; i <= 5; i++)
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failures += test_encode_decode(infile, 29, 1, 0, i, 0, 0); // HE-AACv2 VBR
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}
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if (channels == 1)
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failures += test_encode_decode(infile, 23, 1, 0, 0, 64000, 0); // AAC-LD
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failures += test_encode_decode(infile, 39, 1, 0, 0, 64000, 0); // AAC-ELD
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failures += test_encode_decode(infile, 39, 1, 1, 0, 64000, 0); // AAC-ELD with SBR
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free_encoder_input();
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printf("%d failures\n", failures);
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return failures;
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}
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